Enabling sampling rate diversity in a voice communication system

ABSTRACT

An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing moving spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a divisional application which claims the benefit ofpriority to U.S. National Phase patent application Ser. No. 14/384,350filed Sep. 10, 2014 which is a based on PCT International ApplicationNo. PCT/US2013/033228 filed Mar. 21, 2013 and claimed priority to U.S.Provisional Patent Application Nos. 61/614,582 filed 23 Mar. 2012 and61/625,576 filed 17 Apr. 2012, which are hereby incorporated byreference in their entirety.

TECHNICAL FIELD

The invention disclosed herein generally relates to audio communicationtechnique and more precisely to a digital audio communication systemsupporting endpoints that apply non-uniform sampling rates. It isintended to apply the teachings of the invention to a voice or videoconference network.

BACKGROUND

In an audio communication system with several endpoints (or clients),there is typically a recurring need to provide an audio signal being anadditive mix of live signals originating from different endpoints andapproximating the sound that would have been heard if all thecommunicating parties had been present in one location. This isgenerally desirable in voice conference systems and video conferencesystems. The literature contains descriptions, discussions and solutionsto many problems arising in connection with such mixing, includingreformatting, delay limiting, load reduction, synchronization, qualityof service issues and the like.

The present invention is applicable particularly to communicationsystems with non-uniform sampling rates. An important example is systemswhere individual communication endpoints are allowed to encode and/ordecode audio signals at a selectable sampling rate, such as 8 kHz(narrowband, as per ITU G.711), 16 kHz (wideband) and 32 kHz (superwideband). At critical sampling, this means that the spectral range, inwhich the spectral content is defined and encoded, is allowed to varybetween different audio signals in the audio communication system.Because the sampling rate is allowed to vary, a system entity (e.g.,conference server) that is responsible for generating the mix signal maybe receiving input audio signals at different sample rates.

A straightforward though computationally costly way of enabling themixing as such may be to decode the input audio signals before mixingand then re-encode the mix signal afterwards. Again, however, theendpoints may require different coding formats with different samplingrates, and so the re-encoding process may need to be repeated once foreach unique format and sampling rate. Alternatively, the mixer outputs abasic signal which is then reformatted into whatever formats arerequired by the connected endpoints. Either way, this placesconsiderable load on the server.

BRIEF DESCRIPTION OF THE DRAWINGS

Example embodiments of the invention will now be described withreference to the accompanying drawings, on which:

FIG. 1 shows an audio communication system with three endpoints;

FIG. 2 shows a detail of an audio communication endpoint including apre-processor and a decoder;

FIGS. 3 and 4 illustrate signals arising at three points in the endpointshown in FIG. 2;

FIG. 5 shows a signal processing path in the audio communication systemof FIG. 1;

FIG. 6 illustrates different processing stages in the decoding of anaudio signal based on spectral band replication; and

FIG. 7a and FIG. 7b illustrate the content of a bitstream representingan audio signal encoded by a spectral band replication technique.

All the figures are schematic and generally only show parts which arenecessary in order to elucidate the invention, whereas other parts maybe omitted or merely suggested. Unless otherwise indicated, likereference numerals refer to like parts in different figures.

DESCRIPTION OF EMBODIMENTS I. Overview

It is an object of the present invention to propose methods and devicesfor enabling sampling rate diversity in an audio communication system.It is a further object to propose devices for an audio communicationsystem in which communication nodes exchange audio data streamsconformal to a sampling-rate independent bitstream format. It is afurther object to facilitate mixing of audio streams associated withendpoints that accept audio data at non-uniform sampling rates and/orsupply audio data at non-uniform sampling rates. It is a still furtherobject to propose an audio communication endpoint capable of decoding abitstream containing spectral components relating to a frequency rangethat is variable and configurable by a different communication nodeproducing the bitstream, such as a communication server or a differentendpoint.

Accordingly, example embodiments of the invention provide methods,devices and computer-program products with the features set forth in theindependent claims.

In a first example embodiment, an audio communication endpoint isconfigured to process a received bitstream with spectral componentsrepresenting spectral content of an audio signal. The spectralcomponents relate to a first frequency range extending up to a firstbreak frequency. Hence, frequencies in the first frequency range areassociated with values of corresponding spectral components, while thespectral components above the first frequency—if any—are unassigned. Theaudio communication endpoint is communicatively connected to at leastone further node in an audio communication network. The audiocommunication endpoints, server and network may be collectively referredto as an audio communication system in this disclosure.

The endpoint further includes a decoder for performing inversetransformation on a second range of spectral components and apre-processor which when active adapts the received bitstream by eitherextending or restricting the frequency range for which the spectralcomponents have been assigned values. Preferably, the pre-processor isarranged upstream of the decoder, so that this range extension or rangerestriction is performed on a frequency-domain representation. Thefrequency range for which the spectral components have been assignedvalues is adapted by the pre-processor in such manner that it coincideswith the second frequency range, namely by removing component values(range restriction) or adding component values (range extension) betweenthe upper bounds of the first and second frequency ranges. The addedvalues may be neutral values corresponding to no excitation, such aszeros.

This example embodiment represents an alternative to upsampling ordownsampling in order to change the frequency range (or equivalently,the sampling rate) of a signal. Put differently, this example embodimentproposes devices that communicate with one another in conformity with asampling-rate independent bitstream format.

The inventors have realized, however, that the removal and addition ofspectral component values may introduce a sharp spectral transition thatmay be produce audible artefacts upon inverse transformation, such asdistortion products, pre-ringing and the like. To avoid such drawbacks,according to this example embodiment, the pre-processor is furtheradapted to attenuate spectral content in a neighbourhood of the cut-offfrequency being the upper bound of either the first or second frequencyrange, whichever is lower. Hence, the cut-off frequency is where aspectral discontinuity may be expected to arise as a consequence of therange extension and range restriction.

If performed in the frequency domain, the attenuation may includemultiplying the spectral components near the cut-off frequency by asequence of downscaling factors causing the components to decreasegradually towards the cut-off frequency. The resulting spectral decaymay be complete (i.e., roll-off down to zero) or partial. The spectralcomponents affected by the downscaling may lie in a smaller or largerneighbourhood of the cut-off frequency. The neighbourhood may besingle-sided, located entirely on one side of the cut-off frequency, ordouble-sided, located on both sides of the cut-off frequency.Preferably, the neighbourhood is left-sided, whereby spectral componentsrelating to frequencies in an interval extending up to the cut-offfrequency are affected by the attenuation.

Alternatively, the attenuation is performed in the time-domain, that is,after the audio signal has been reconstructed by way of the inversetransformation. The attenuation may be carried out by a low-pass filter,preferably one having a magnitude that falls off smoothly between thepass band and the stop band of the filter. Efficient analogue anddigital implementations of low-pass filters are well known in the art.

In a further development of the first example embodiment, theattenuation is conditional upon characteristics of the spectral decay,that is, properties relative to the fall-off behaviour of spectralcomponents pertaining to frequencies near the cut-off frequency. Indeed,as the inventors have realized, if spectral attenuation can be dispensedwith, it is preferably omitted to reduce the risk of introducing newartefacts. In particular, duplicated low-pass filtrations will degradethe signal content, e.g., in terms of signal-to-noise ratio since thetotal amplitude is locally attenuated. As will be explained in whatfollows, cases where attenuation can be dispensed with typically arisewhen the pre-processor adds spectral components (range extension). Suchsituations typically do not arise in connection with frequency rangerestriction. To this end, the pre-processor is configured tocharacterize the spectral decay of the adapted received bitstream, thatis, after the bitstream has undergone removal or addition of spectralcomponents. It is envisaged that the pre-processor may characterize thespectral decay either by a direct appraisal or by considering anindirect indicator. A direct appraisal may include a search fordiscontinuities in the spectral component values or for segments withstrong local variation. If a discontinuity (or strong local variation)is found, it may be expected that the spectral decay is not gradual. Asthe skilled person will realize, the notions of ‘discontinuity’ and‘strong variation’—corresponding to the cases requiring correctiveaction—may be quantified by listening experiments. A direct appraisalmay further include estimating a local spectral decay rate on the basisof the spectral components.

Among indirect indicators, the processing history of an audio signal mayallow conclusions as to whether the spectral decay is sufficientlygradual or whether it needs further attenuation. To this end, thebitstream may include an indicator evidencing that the spectralcomponents or the underlying time representation of the signal hasundergone a processing step that is one of low-pass filtering, spectralshaping, pre-sampling filtering (aiming to reduce the impact ofaliasing) or other operations ensuring or contributing to a gradualspectral decay. The pre-processor may then be configured to read a valueof the indicator and to conclude, based on the nature of the processingindicated, whether to carry out attenuation of spectral content or not.

An audio communication endpoint may further include an interface facingaway from any lines connecting the endpoint to other nodes in the audiocommunication network. Such interface may be regarded as an outerboundary point of the audio communication network. The interface may bea user interface with transducers for outputting reproduced speech (andinputting natural speech), e.g., speakers (and microphones).Alternatively, it may be a network interface allowing the speech to betransmitted (or received) in encoded form over a network or transmissionline after the processing by the endpoint is complete; transducersallowing user interaction may be arranged at the far end of the networkor transmission line. In particular, the interface may act as a bridgeto a public switched telephone network.

In a second example embodiment, the received bitstream contains anenergy envelope relating to the full first frequency range. Some of thespectral components relating to the first frequency range are encoded byspectral band replication (SBR). Then, as is known per se in the art, acore range included in the first frequency range comprises spectralcomponents with explicit values. The core range may be a subinterval ofthe first frequency range that lies below a cross-over frequency.Further, spectral components relating to frequencies in the firstfrequency range but outside the core range are derivable bytransposition of the spectral components in the core range, wherein theenergy envelope indicates the correct scale of the spectral componentvalues obtained by transposition. In accordance with this exampleembodiment, the spectral components relating to frequencies in the firstfrequency range but outside the core range are derived prior to theattenuation of spectral content in a neighbourhood of the cut-offfrequency. Addition of neutral-valued spectral components may beperformed either before or after the attenuation, with no known impacton the result.

In a third example embodiment, an audio communication endpoint comprisesan encoder and a transcoder. The communication endpoint may further haveone or more of the features discussed in connection with the precedingembodiments, e.g. decoder, pre-processor and the like. However, thecommunication endpoint may also be a pure input node to the audiocommunication network. In this example embodiment, the encoder encodesan outgoing audio signal by means of a transform sup-plying spectralcomponents relating to a third frequency range. The transcoder receivesthe spectral components from the encoder and outputs a bitstream that isconformal to a given bitstream format by which spectral componentsrelating to frequencies up to a maximum frequency can be transmitted. Inthe bitstream, the transcoder encodes spectral components in the thirdfrequency range and leaves spectral components relating to higherfrequencies—if the bitstream format allows spectral components in ahigher range—unassigned. This way, the audio communication endpoint willbe able to communicate, by way of the bitstream, with other nodes in theaudio communication network, e.g., with a server performing mixing andwith different audio communication endpoints. Indeed, othercommunication endpoints in the audio communication network are able toprocess a received bitstream even if this contains spectral componentslying outside the frequency ranges on which the decoders in theseendpoints operate or if the spectral components in the bitstream do notcompletely fill the frequency range operated on.

In a further development of the third example embodiment, the endpointfurther comprises a filter operable to attenuate spectral content in aneighbourhood of a third break frequency being the upper bound of thethird frequency range. The filter may be a pre-sampling filter arrangedupstream of the encoder, e.g., a low-pass filter. Alternatively, thefilter may be a frequency-domain filter arranged downstream of theencoder. The transcoder is adapted to detect a condition of the filter(e.g., enabled, disabled) and assign a value to a pre-filtering field inthe bitstream output from the endpoint, wherein the value is inaccordance with the detected condition. Preferably, the pre-filteringfield contains a value of the third break frequency. As discussed above,this may simplify subsequent processing of the bitstream and/or improvethe quality of a final output.

In an example embodiment, an audio communication server is configured tosend bitstreams to audio communication endpoints and to receivebitstreams from these endpoints. Each of the bitstreams may containspectral components representing spectral content of an audio signal andis conformal to a predefined bitstream format allowing transmission ofspectral components up to a maximum frequency. The endpoints may haveproperties similar to those described above. In particular, the spectralcomponents in each bitstream received by the server (incoming bitstream)relate to a frequency range extending up to an input break frequencywhich is selectable by the corresponding audio communication endpoint.The number of endpoints may be three or more, whereby a given endpointmay require a mix of signals originating from two or more otherendpoints. The server may be operable to output one outgoing bitstream,whereby all endpoints receive a common signal (e.g., by broadcasttransmission over the network) informing them of the content of theongoing audio communication. Alternatively, the server is operable tooutput a plurality of different outgoing bitstream (e.g., by unicasttransmission over the network), possibly one for each receivingendpoint.

In this example embodiment, the audio communication server comprises amixer and a selector. The selector controls the output of the server (orone of the outputs, if the server provides more than one output) insofaras the output is either a mix signal provided by the mixer or a signalthat reproduces one of the inputs. (In an analogue signal processingsituation similar to the present one, this may amount to forwarding thesignal without substantive processing, e.g., after a mereamplification.) In the latter case, the outgoing bitstream may be abitstream reproducing one of the incoming bitstreams. The outgoingbitstream may optionally undergo frequency range extension or frequencyrange restriction, as discussed above, so as to correspond to an outputbreak frequency that the server is expected to supply to the endpointsin the system. Preferably, the mixer is configured to supply an outgoingbitstream produced in this manner in time segments where only one of theincoming bitstreams is active. Activity of an incoming bitstream may beascertained by performing voice activity detection on the bitstreams;alternatively, the endpoint from which the bitstreams originate maysupply metadata indicating the points in time at which audio activitybegins and ends. Both the selector decision and the reproduction of theincoming bitstream may be accomplished without knowledge of the inputbreak frequency (or sampling rate) of the incoming bitstream. This ismade possible by the adaptability of the endpoints, i.e., their abilityto handle any outgoing bitstreams from the server that are conformalwith the bitstream format.

In one example embodiment, which may either be a further development ofthe preceding embodiment or practised on its own, the server adapts thebreak frequency (or mixer break frequency) in an outgoing bitstream inaccordance with properties of the audio communication endpointsreceiving the outgoing bitstream. For instance, the server may receivean output break frequency defining a frequency range on which a givenaudio communication endpoint operates to reconstruct an audio signalencoded by an outgoing bitstream. Gathering the output break frequenciesfrom all endpoints, the server may be able to conclude that the mixerbreak frequency can be set to a smaller value than the maximum breakfrequency permitted by the predefined bitstream format. For instance,the mixer break frequency may be set to the maximum among the outputbreak frequencies of the endpoints. This reduces the computational loadon the server.

The dependent claims define example embodiments of the invention, whichare described in greater detail below. It is noted that the inventionrelates to all combinations of features, even if the features arerecited in different claims.

II. Example Embodiments

FIG. 1 is a generalized block diagram of an audio communication system100 with one server 190 and three endpoints 110, 120, 130. In thisexample embodiment, each endpoint 110, 120, 130 comprises a microphone115, 125, 135, a speaker 116, 126, 136 and associated processing means111, 112, 121, 122, 131, 132. The first and second endpoints 110, 120are connected to the server 190 via respective communication lines 119,129 in an audio communication network. The third endpoint 130 isdistributed spatially, wherein the processing means 131, 132 arearranged in a first portion 130 a acting as interface between the audiocommunication network 138 (which may be a packet-switched network) onits left-hand side in the figure and a public switched telephone network(PSTN) 139 on its right-hand side. The PSTN 139 connect the firstportion 130 a to a second portion 130 b, in which the microphone 135 andspeaker 136 are arranged.

The endpoints 110, 120, 130 operate at different sampling rates. Forinstance, the third endpoint 130 associated with the PSTN 139 mayoperate at narrowband rate (8 kHz), while the first and second endpoints110, 120 may operate at wideband or super wideband rates. Still,bitstreams are transmitted over the audio communication network in auniform bitstream format. The bitstream format accommodates spectralcomponents extending from a predefined minimum frequency (e.g., 20 Hz or0 Hz) up to a variable break frequency, which may have any value betweenthe minimum frequency and a predefined maximum frequency (e.g., 20 000Hz) specified for the bitstream format. At critical sampling, thesampling rate is approximately equal to twice the first break frequency.The bitstream format allows the spectral components relating tofrequencies up to the first break frequency to carry values. Spectralcomponents relating to higher frequencies are unassigned. In thisconnection, it is advantageous to apply some type of entropy encoding(e.g., Huffman coding), by which the presence of unassigned spectralcomponents in the bitstream occupies a limited amount of additionalbandwidth in the audio communication network. The invention does notpresuppose use of any particular transform; as one of many possibleoptions, it may use a harmonic discrete transform with overlapping timewindows and a time stride of the order of tens of milliseconds; thetransform may be MDCT or DCT.

In the example embodiment, the bitstream format allows the first breakfrequency to be one of the predefined values 4 kHz, 8 kHz and 16 kHz,corresponding to sampling rates of about 8 kHz, 16 kHz and 32 kHz,respectively. Similarly, the endpoints 110, 120, 130 apply a secondbreak frequency that is one of these three predefined frequencies.Hence, the first and second frequency ranges are unions of the frequencysubbands [0, 4], [4, 8] and [8, 16] (unit: 1 kHz). The widths of thesubbands are 1:1:2. This is however not an essential feature of thisexample embodiment of the invention.

In normal operation, the server 190 is configured to receive incomingbitstreams from each of the endpoints 110, 120, 130 and to generate amix signal obtained by additive mixing of the signals represented by theincoming bitstreams. To achieve this, a mixer 192 within the server 190decodes the incoming bitstreams partially or completely, in such mannerthat the spectral components representing audio signals originating fromeach endpoint 110, 120, 130 become available and can be operated on. Asone example, the mixing may be additive and frequency bin-wise.

The mixer 192 may be configured to produce other combinations than a mixof all incoming signals. For instance, the mixer may be connected to theendpoints 110, 120, 130 via individual outgoing lines (not shown), sothat a signal specifically adapted is supplied to each particularendpoint. If the mixer 192 is adapted to produce plural output signals,it may be desirable to exclude an incoming signal from the m^(th)endpoint from an output signal intended in particular for the m^(th)endpoint; this may be perceptually more comfortable for a user and mayalso reduce the likelihood of feedback instability.

The server 190 is configured to output the mix signal as a bitstream inaccordance with the network-wide bitstream format, wherein it may setthe first break frequency to any of 4, 8 and 16 kHz. The selection of afirst break frequency value may depend on available computationalresources in the server 190, network bandwidth, computational resourcesavailable for decoding in the endpoints 110, 120, 130, sampling rate ofthe incoming bitstreams and the like. However, because all endpoints110, 120, 130 are able to decode bitstreams having an arbitrary one ofthese sampling rates, the server 190 it is not strictly required toadapt the signal to the second break frequency applied by particularendpoints. Indeed, if all endpoints 110, 120, 130 apply 4 kHz as secondbreak frequency, there is typically little point in outputting awideband or super wideband mix signal; if however there is diversityamong different endpoints 110, 120, 130, then the individualization ofthe bitstreams is preferably handled on the endpoint side rather than onthe server side.

In the example embodiment, the server 190 is configured to simplify themixing in cases where only a single incoming bitstream is active (e.g.,in terms of voice activity). In such cases, as symbolically illustratedby the selector 191, the server 190 may be configured to forward anunprocessed incoming bitstream as output. Indeed, in its upper positionon the drawing, the selector 190 joins a switch 193 that forwards aselected one the incoming bitstreams on the one hand and the outputpoint of the server 190 on the other hand. Preferably, the switch 193 isadapted to forward the active bitstream in case only one bitstream isactive. The functionality of this operational mode is made possible byvirtue of the adaptability of the endpoints 110, 120, 130, which asalready discussed may decode a bitstream from the server 190 regardlessof its sampling rate. Because the mixer 192 can be disabled while theselector 191 is in its upper position (bypass position), thisoperational mode implies a potential saving in computational load.

FIG. 2 illustrates details of the processing means 112 in an endpoint110. The processing means 112 includes a pre-processor 201, a decoder202 and an optional unit 203 responsible for parsing a pre-filteringfield in the bitstream and forward this to the pre-processor 201.(Alternatively, the pre-processor 201 performs the parsing operationinternally and the parsing unit 203 is omitted.)

FIG. 3 contains an example of the signals arising in the processingmeans 112 during operation. Signal A illustrates the signal componentsin the received bitstream, out of which spectral components relating tofrequencies up to a first break frequency f1 are assigned andhigher-frequency spectral components are unassigned. Signal Billustrates the spectral components after the bitstream has been adaptedin accordance with the second break frequency f2 applied by thisendpoint 110. The spectral components in the range [f1, f2] have beenassigned neutral values. Because the first break frequency f1 isrelatively lower, it acts as cut-off frequency f0. The pre-processor 201is adapted to attenuate spectral content in a neighbourhood of thecut-off frequency f0. FIG. 3 illustrates two possible neighbourhoodsthat may be selected for this purpose, namely a double-sidedneighbourhood J1 and a single-sided neighbourhood J2. After theattenuation of the adapted received bitstream, the decoder 202 performsinverse transformation by operating on the spectrum components up to thesecond break frequency f2, whereby a time-domain representation of thesignal is obtained, as illustrated by signal C in FIG. 3.

FIG. 2 illustrates a processing means 112 in which the spectralattenuation is performed on a frequency-domain representation of thesignal. In the figure, there is further indicated an optional low-passfilter 204 arranged downstream of the decoder 202. The low-pass filter204 is operable to attenuate spectral content in a neighbourhood of thecut-off frequency by operating on a time-domain representation of thesignal. The low-pass filter 204 may be controlled by an output from theparsing unit 203, similarly to the pre-processor 201. Hence, theprocessing of the incoming signal is carried out in a distributedfashion: the pre-processor 201 performs range adaptation, the decoder202 performs inverse transformation, and the low-pass filter 204performs spectral attenuation. Alternatively, if direct estimation ofthe spectral decay is used, the pre-processor 201 performs the rangeadaptation and estimates the spectral decay; it then communicates to thelow-pass filter 204, downstream of the decoder 202, whether the spectraldecay is already sufficiently gradual or if spectral attenuation is tobe activated.

FIG. 4 illustrates a case where the second break frequency f2 is lessthan the first break frequency f1 and will therefore act as cut-offfrequency, see signals A′ and B′, respectively extracted from locationsA and B in the circuit of FIG. 2. The pre-processor 201 then attenuatesspectral content in a neighbourhood of the cut-off frequency, such as inthe frequency interval J3 in FIG. 4.

FIG. 5 illustrates how an example embodiment of the invention avoidsprocessing that would duplicate spectral attenuation, which mayotherwise have a negative impact on the final signal quality. Thisfigure shows a data path from an input microphone 115 in the firstendpoint 110, via processing means 111 in the same endpoint, via theserver 190, via processing means 122 in the second endpoint 120 and upto a speaker 126 arranged in the second endpoint 120. Spectralcoefficients representing the signal obtained at the microphone 115 areobtained by an MDCT stage 512. In connection with this, there is ananti-aliasing filter 511, 513 located either upstream of the MDCT stage512 (whereby the filtering proceeds in the time domain) or downstream ofthe MDCT stage 512 (whereby the filtering proceeds in the frequencydomain). The anti-aliasing filter 511, 513 may have a magnitude responsewith respect to frequency as indicated in the figure, wherein the stopband extends from the first break frequency f1 and upwards. The presentinvention does not place any particular requirements on thecharacteristics of these anti-aliasing filters. However, when the outputsignal from the processing means 111 is to be transmitted over a givenPSTN, it may be preferable to select a filter corresponding closely toany specifications for that PSTN, as this may ensure optimal quality inthe circumstances. In the processing means 111, further, a multiplexer514 produces a bitstream intended as final output to be transmitted tothe server 190. The bitstream may include an indication that the signalhas undergone anti-aliasing filtering. Optionally, the indicationincludes the value of the first break frequency f1. The indication maybe localized in a pre-filtering field defined in the network-widebitstream format.

In this example, the server 190 does not process the bitstream from thefirst endpoint 110 any further. Alternatively, the server 190 processesthe bitstream in such manner that the value assigned to thepre-filtering field is conveyed to the downstream side.

In the processing means 122 within the second endpoint 120, apre-processor 522 adapts the bitstream by adding neutral-valued spectralcomponents, so that the assigned range matches the second breakfrequency that the second endpoint 120 applies. In order to fulfil itsduties in the system, the processing means 122 does not necessarily haveaccess to information concerning the spectral decay of the signal, thatis, whether it is gradual or abrupt. To handle signals possibly havingabrupt spectral decay, the processing means 122 is equipped with alow-pass filter 523, which is located upstream of an inverse MDCT stage525 and which can be included in the signal processing path by actuatinga selector 524, symbolically illustrated by a simple switch in FIG. 5.For the purpose of deciding whether to include the low-pass filter 523in the processing path or not, a demultiplexer 521 extracts the value ofthe pre-filtering field in the bitstream and forwards the value to theselector 524. The value of the pre-filtering field may provide indirectinformation about the spectral delay. In the present case, it may bedecided to omit the low-pass filtration in the second endpoint 120,since it is known that anti-aliasing filtration has been carried out inthe first endpoint 110. This operation has probably ensured that thespectral decay is sufficiently gradual in order for noticeable artefactsnot to arise.

FIG. 5 also suggests an alternative location for a low-pass filter 526,namely downstream of the inverse MDCT stage 525. The low-pass filter 526in this position operates on a time-domain representation of the signal.Similar to the low-pass filter 523 in its first location, thetime-domain low-pass filter 526 can be enabled and disabled inaccordance with the value of a pre-filtering field in the bitstream. Thevalue of the pre-filtering field can be extracted by the demultiplexer521 and provided either directly to the low-pass filter 526 or toselection means controlling whether or not the low-pass filter 526 is toform part of the signal processing path. Hence, both in the receivingand the sending endpoint, the low-pass filtering can be carried outeither in the time domain or the frequency domain. In the exampleembodiment shown in FIG. 5, there are no particular requirements on thecharacteristics of these low-pass filters 523, 526. However,considerations may be needed in an alternative example embodiment,wherein the processing means 122 is included in a spatially distributedendpoint similar to that endpoint 130 in FIG. 1 which is connected to aPSTN 139. Indeed, when the output signal is to be transmitted over agiven PSTN, it may be preferable to occupy the relevant one of thefilter positions 523, 526 in FIG. 5 by a filter which is in agreementwith any specifications for that PSTN, as this may ensure optimalquality in the circumstances.

In a case where information concerning the processing history of theincoming bitstreams is available, the decision whether to apply spectralattenuation or not may be guided by rule of the type expressed in Table1 below.

TABLE 1 Output 8 kHz Output 16 kHz Output 32 kHz Input 8 kHz (1, —, —)(F, 0, —) (F, 0, 0) Input 8 kHz (shaped) (1, —, —) (1, 0, —) (1, 0, 0)Input 16 kHz (F, X, —) (1, 1, —) (1, F, 0) Input 16 kHz (shaped) (F, X,—) (1, 1, —) (1, 1, 0) Input 32 kHz (F, X, X) (1, F, X) (1, 1, 1) Input32 kHz (shaped) (F, X, X) (1, F, X) (1, 1, 1)The entries in the table are triples (a, b, c), where a refers to theprocessing or content of the lower [0, 4] subband, b refers to theprocessing or content of the centre [4, 8] subband, and c similarly tothe top [8, 16] subband. The notation has the meaning indicated in Table2 below.

TABLE 2 0 Present in output, obtained by padding with neutral values XAbsent in output, removed from input — Absent in output and absent ininput F Present in output and in input; spectral attenuation applied 1Present in output and in inputThe cases where a signal is decoded without a change in sampling rateare straightforward. In decoding to a lower sampling rate, spectralattenuation is applied in the highest active subband. In decoding to ahigher sampling rate, spectral attenuation is applied in the highestactive subband except where it is known that the input has alreadyundergone similar processing, e.g., spectrum shaping, anti-aliasingfiltering, low-pass filtering, pre-filtering, as indicated by “shaped”.For example, the case of decoding a shaped 16 kHz signal at 32 kHz doesnot require spectral attenuation, since there is already a soft roll-offin the centre subband. When the same signal is decoded at 8 kHz,however, the spectrum shaping, which was applied to the centre band, isof no use since the lower subband will be the highest one in the signalto be decoded; for this reason, spectral attenuation is preferablyapplied to this signal before it undergoes inverse transformation.

With reference now to FIGS. 6, 7 a and 7 b, an example will be discussedin which the present invention is combined with SBR. FIG. 6 illustratesthree frequency-domain representations, corresponding to differentprocessing stages in an audio communication system endpoint configuredto process a bitstream for the purpose of outputting an audio signal viaa speaker or the like. In FIG. 6, a core range extends from the minimumfrequency up to a cross-over frequency f_(co). Here, the spectralcomponents have values assigned. The bitstream contains these values aswell as an energy envelope relating to the full first frequency range.The first frequency range continues from the cross-over frequency f_(co)up to the first break frequency f1, in which range the spectralcomponents are not known exactly but may be reconstructed by transposingthe component value from the core range and scaling them in theirtransposed positions in accordance with the energy envelope. FIG. 7a andFIG. 7b illustrates the SBR technique in more detail, in a case wherethe cross-over frequency f_(co)≈3 200 Hz and the first break frequencyf1≈8 000 Hz. That is, FIG. 7a shows MDCT spectrum coefficients and acorresponding energy envelope before encoding. The envelope isrepresented at lower resolution (with respect to frequency) than theMDCT spectrum coefficients. In this example, the envelope is piecewiseconstant by segments of about 300 Hz and then, from about 4 000 Hzonwards, varies by segments of about 600 Hz each. On the other hand,FIG. 7b , showing the same signal after SBR encoding, the MDCT spectrumcoefficients above the cross-over frequency have been removed, which mayreduce the bitrate down to about 50% of its original value.

Returning to FIG. 6, the upper portion illustrates the signal asextracted from the bitstream, wherein only spectral components in thecore range have been assigned values. The second portion of FIG. 6illustrates the SBR reconstruction process, wherein spectral componentsfrom the cross-over frequency f_(co) up to the first break frequency f1are assigned their values by transposition of values of spectralcomponents relating to frequencies below the cross-over frequencyf_(co). The transposition, which may for instance be of a copy-up,single-sideband or harmonic type, is preferably accompanied by rescalingin accordance with the energy envelope (not shown) extracted from thebitstream. To obtain the signal illustrated by the lower portion of FIG.6, a zero-padding operation similar to the one described in connectionwith the preceding example embodiments is applied.

If the signal has been obtained by sampling preceded by low-passfiltering (as may be explicitly encoded in a field in the bitstreamformat to facilitate a decision not to attenuate), it may be expectedthat the signal illustrated in the lowest portion may in this caseproceed to inverse transformation without any preliminary spectralattenuation. Indeed, even though the bitstream does not carry explicitvalues of the spectral components in a neighbourhood of the first breakfrequency f1 (which plays the role of cut-off frequency f0), thespectral decay is conveyed by the energy envelope. It is noted that thisis likely not the case for the signal shown in FIG. 7 and FIG. 7b ,since the spectrum has a substantially constant, moderately high valuefrom about 6 000 Hz onwards. Hence, it may be appropriate to attenuatethe spectral content in this range to make it gradually decay towards 8000 Hz before the signal undergoes inverse transformation.

III. Equivalents, Extensions, Alternatives and Miscellaneous

Further embodiments of the present invention will become apparent to aperson skilled in the art after studying the description above. Eventhough the present description and drawings disclose embodiments andexamples, the invention is not restricted to these specific examples.Numerous modifications and variations can be made without departing fromthe scope of the present invention, which is defined by the accompanyingclaims. Any reference signs appearing in the claims are not to beunderstood as limiting their scope.

The systems and methods disclosed hereinabove may be implemented assoftware, firmware, hardware or a combination thereof. In a hardwareimplementation, the division of tasks between functional units referredto in the above description does not necessarily correspond to thedivision into physical units; to the contrary, one physical componentmay have multiple functionalities, and one task may be carried out byseveral physical components in cooperation. Certain components or allcomponents may be implemented as software executed by a digital signalprocessor or microprocessor, or be implemented as hardware or as anapplication-specific integrated circuit. Such software may bedistributed on computer readable media, which may comprise computerstorage media (or non-transitory media) and communication media (ortransitory media). As is well known to a person skilled in the art, theterm computer storage media includes both volatile and nonvolatile,removable and non-removable media implemented in any method ortechnology for storage of information such as computer readableinstructions, data structures, program modules or other data. Computerstorage media includes, but is not limited to, RAM, ROM, EEPROM, flashmemory or other memory technology, CD-ROM, digital versatile disks (DVD)or other optical disk storage, magnetic cassettes, magnetic tape,magnetic disk storage or other magnetic storage devices, or any othermedium which can be used to store the desired information and which canbe accessed by a computer. Further, it is well known to the skilledperson that communication media typically embodies computer readableinstructions, data structures, program modules or other data in amodulated data signal such as a carrier wave or other transportmechanism and includes any information delivery media.

The invention claimed is:
 1. A audio communication server for exchangingbitstreams with a plurality of audio communication endpoints, each ofsaid bitstreams containing spectral components representing spectralcontent of an audio signal and conformal to a predefined bitstreamformat allowing transmission of spectral components up to a maximumfrequency, wherein the spectral components in each incoming bitstreamrelate to a frequency range extending up to an input break frequencywhich is selectable by each corresponding audio communication endpoint,said server comprising: a mixer configured to receive a plurality ofincoming bitstreams and to output, based thereon, a bitstreamrepresenting an audio signal being an additive mix of at least one ofthe incoming bitstreams; and a selector configured to output, from theaudio communication server, an outgoing bitstream being either abitstream output by the mixer or a bitstream reproducing an active oneof the incoming bitstreams, wherein the selector is configured tomonitor the incoming bitstreams for audio activity and to output, inresponse to having exactly one active incoming bitstream, an outgoingbitstream reproducing the active incoming bitstream.
 2. The audiocommunication server of claim 1, wherein the mixer is further configuredto: receive, from each audio communication endpoint, an output breakfrequency defining a frequency range on which the audio communicationendpoint operates to reconstruct an audio signal encoded by an outgoingbitstream; and output a bitstream containing spectral componentsrelating only to a frequency range extending up to an mixer breakfrequency being less than or equal to the least of the output breakfrequencies received from the audio communication endpoints.
 3. Theaudio communication server of claim 1, wherein the plurality of audiocommunication endpoints includes at least three audio communicationendpoints.
 4. The audio communication server of claim 1, wherein themixer is configured to partially decode the plurality of incomingbitstreams.
 5. The audio communication server of claim 1, wherein themixer is configured to completely decode the plurality of incomingbitstreams.
 6. The audio communication server of claim 1, wherein themixer is configured to perform mixing that is additive and frequencybin-wise.
 7. The audio communication server of claim 1, wherein themixer is configured to exclude an incoming bitstream of the plurality ofincoming bitstreams to reduce a likelihood of feedback instability. 8.The audio communication server of claim 1, wherein the mixer isconfigured to be disabled when the selector outputs the bitstreamreproducing the active one of the incoming bitstreams.
 9. The audiocommunication server of claim 1, wherein the output break frequency foran audio communication endpoint of the plurality of audio communicationendpoints is one of 4 kHz, 8 kHz and 16 kHz.
 10. A method of operatingan audio communication server for exchanging bitstreams with a pluralityof audio communication endpoints, each of said bitstreams containingspectral components representing spectral content of an audio signal andconformal to a predefined bitstream format allowing transmission ofspectral components up to a maximum frequency, wherein the spectralcomponents in each incoming bitstream relate to a frequency rangeextending up to an input break frequency which is selectable by eachcorresponding audio communication endpoint, wherein the audiocommunication server comprises a mixer and a selector, said methodcomprising: receiving, by the mixer, a plurality of incoming bitstreamsand outputting based thereon, by the mixer, a bitstream representing anaudio signal being an additive mix of at least one of the incomingbitstreams; and outputting from the audio communication server, by theselector, an outgoing bitstream being either a bitstream output by themixer or a bitstream reproducing an active one of the incomingbitstreams, the method further comprising: monitoring, by the selector,the incoming bitstreams for audio activity and to outputting by theselector, in response to having exactly one active incoming bitstream,an outgoing bitstream reproducing the active incoming bitstream.
 11. Themethod of claim 10, further comprising: receiving, by the mixer fromeach audio communication endpoint, an output break frequency defining afrequency range on which the audio communication endpoint operates toreconstruct an audio signal encoded by an outgoing bitstream; andoutputting, by the mixer, a bitstream containing spectral componentsrelating only to a frequency range extending up to an mixer breakfrequency being less than or equal to the least of the output breakfrequencies received from the audio communication endpoints.
 12. Themethod of claim 10, wherein the plurality of audio communicationendpoints includes at least three audio communication endpoints.
 13. Themethod of claim 10, further comprising: partially decoding, by themixer, the plurality of incoming bitstreams.
 14. The method of claim 10,further comprising: completely decoding, by the mixer, the plurality ofincoming bitstreams.
 15. The method of claim 10, further comprising:performing mixing, by the mixer, that is additive and frequencybin-wise.
 16. The method of claim 10, further comprising: excluding, bythe mixer, an incoming bitstream of the plurality of incoming bitstreamsto reduce a likelihood of feedback instability.
 17. The method of claim10, further comprising: disabling the mixer when the selector outputsthe bitstream reproducing the active one of the incoming bitstreams. 18.The method of claim 11, wherein the output break frequency for an audiocommunication endpoint of the plurality of audio communication endpointsis one of 4 kHz, 8 kHz and 16 kHz.